[asterisk-bugs] [Asterisk 0014249]: One way voice after attended transfer

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14249 
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Reported By:                RadicAlish
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   14249
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.21.2 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-01-15 04:49 CST
Last Modified:              2009-01-23 14:19 CST
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Summary:                    One way voice after attended transfer
Description: 
We have problem with one way voice in next scenario:
101 calls to 102, 102 takes second line and calls to IVR (number 1234)
that have Answer() application and then call directed to 103
102 makes attended transfer while 103 is ringing, so now we have call
between 101 (connected) and 103 (ringing)
when 103 answers we get in CLI bulk of messages:
[15 12:33:10] WARNING[3681] chan_sip.c: Asked to transmit frame type 64,
while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x
8 (alaw)(8)
101 can hear 103, but 103 can't
only if 101 press any digit (sends DTMF) or press HOLD and UNHOLD, then we
have two way voice.

101, 102 and 103 are SIP devices
I attached full log with SIP debug
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---------------------------------------------------------------------- 
 (0098611) svnbot (reporter) - 2009-01-23 14:19
 http://bugs.digium.com/view.php?id=14249#c98611 
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Repository: asterisk
Revision: 170659

_U  branches/1.6.0/
U   branches/1.6.0/main/channel.c

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r170659 | file | 2009-01-23 14:19:14 -0600 (Fri, 23 Jan 2009) | 18 lines

Merged revisions 170652 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | 11 lines
  
  Merged revisions 170648 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4
lines
    
    When a channel is answered make sure any indications currently playing
stop. Usually the phone would do this but if the channel was already
answered then they are being generated by Asterisk and we darn well need to
stop them.
    (closes issue http://bugs.digium.com/view.php?id=14249)
    Reported by: RadicAlish
  ........
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http://svn.digium.com/view/asterisk?view=rev&revision=170659 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-23 14:19 svnbot         Checkin                                      
2009-01-23 14:19 svnbot         Note Added: 0098611                          
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